Field of the Disclosure
The technology of the disclosure relates generally to Web Real-Time Communications (WebRTC) interactive flows.
Technical Background
Web Real-Time Communications (WebRTC) is an ongoing effort to develop industry standards for integrating real-time communications functionality into web clients, such as web browsers, to enable direct interaction with other web clients. This real-time communications functionality is accessible by web developers via standard markup tags, such as those provided by version 5 of the Hypertext Markup Language (HTML5), and client-side scripting Application Programming Interfaces (APIs) such as JavaScript APIs. More information regarding WebRTC may be found in “WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web,” by Alan B. Johnston and Daniel C. Burnett, 2nd Edition (2013 Digital Codex LLC), which is incorporated in its entirety herein by reference.
WebRTC provides built-in capabilities for establishing real-time video, audio, and/or data streams in both point-to-point interactive flows and multi-party interactive flows. The WebRTC standards are currently under joint development by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). Information on the current state of WebRTC standards can be found at, e.g., http://www.w3c.org and http://www.ietf.org.
To establish a WebRTC interactive flow (e.g., a real-time video, audio, and/or data exchange), two WebRTC clients may retrieve WebRTC-enabled web applications, such as HTML5/JavaScript web applications, from a web application server. Through the web applications, the two WebRTC clients then engage in dialogue for initiating a peer connection over which the WebRTC interactive flow will pass. This dialogue may include a media negotiation established to communicate and reach an agreement regarding parameters that define initial characteristics of the WebRTC interactive flow.
In some embodiments, the media negotiation may be implemented via a WebRTC offer/answer exchange. A WebRTC offer/answer exchange typically occurs via a secure network connection such as a Hyper Text Transfer Protocol Secure (HTTPS) connection or a Secure WebSockets connection. In a WebRTC offer/answer exchange, a first WebRTC client on a sender computing device sends an “offer” to a second WebRTC client on a recipient computing device. The offer includes a WebRTC session description object that specifies media types and capabilities that the first WebRTC client supports and prefers for use in the WebRTC interactive flow. The second WebRTC client then responds with a WebRTC session description object “answer” that indicates which of the offered media types and capabilities are supported and acceptable by the second WebRTC client for the WebRTC interactive flow. It is to be understood that the dialogue may employ mechanisms other than a WebRTC offer/answer exchange to establish a WebRTC interactive flow between the WebRTC clients. Once the dialogue is complete, the WebRTC clients may then establish a direct peer connection with one another, and may begin an exchange of media and/or data packets transporting real-time communications. The peer connection between the WebRTC clients typically employs the Secure Real-time Transport Protocol (SRTP) to transport real-time media flows, and may utilize various other protocols for real-time data interchange.
WebRTC includes mechanisms for enabling WebRTC clients to obtain detailed information regarding various Quality of Service (QoS) characteristics of WebRTC interactive flows. These QoS characteristics may be used by a WebRTC client to dynamically monitor and adjust the handling of a WebRTC interactive flow. For example, in response to a decrease in network throughput, a WebRTC client may be able to compensate by adjusting a video or audio resolution, or by selecting a more appropriate codec for encoding the WebRTC interactive flow. However, these QoS characteristics may not be accessible outside the WebRTC client, and thus may not be available to external entities (e.g., enterprises) to monitor, characterize, and optimize a network over which the WebRTC interactive flow passes.